The invention relates to a method and a system for integration of a packet-oriented network into a communication system.
Contemporary communication systems are characterized by the convergence of information and communications infrastructures. In such cases communication systems are known which support voice and/or video communication via packet-oriented networks. This type of system is known mostly in technical circles by the generic term of ‘Voice over IP’ or is abbreviated to ‘VoIP’, with the abbreviation ‘IP’ standing for ‘Internet Protocol’ a generally widely-used protocol for exchange of data over packet-oriented networks. For VoIP communication the H.323 or SIP (Session Initiation Protocol) protocols are widely used.
SIP is a signaling protocol for Internet telephony as well as for other services such as conference interactions, event notification, message switching etc. This protocol was developed by the Working Group MMUSIC (Multiparty Multimedia Session Control) of the Working Group IETF (Internet Engineering Task Force). The H.323 Standard is an international ITU-T (International Telecommunication Union—Telecommunications Standardization Sector) standard for voice, data and video communication over packet-switching networks.
As well as an infrastructure for exchanging packet-oriented data, a telephony infrastructure with fixed lines to communication terminals continues to be widely used. For economic and security reasons therefore there is in many systems no provision for any wholesale changeover to a purely packet-oriented architecture. Despite this, existing communication systems with a fixed line-based telephony infrastructure have been extended by devices for connection of communication terminals based on a packet-oriented network. These types of communication systems are thus often referred to as ‘convergent’ communication systems, which indicates a convergence of the two types of infrastructure.
In communication systems with a telephony infrastructure based on fixed line assignments the communication terminals are controlled by a central communication device.
The central communication device makes service features available and transfers information between communication system-internal or external communication terminals.
By contrast, in the architecture of purely packet-oriented VoIP communication systems there is no provision for central switching and control of communication partners, but instead the control of switching and service features in purely VoIP communication systems is undertaken decentrally in the communication terminals or in distributed control server systems.
Thus for a comprehensive integration of VoIP-based communication terminals into a convergent communication system a purely bilateral conversion of packet-oriented and streamed user information—i.e. voice or video data—is not sufficient. In addition a convergent communication system requires a conversion of signaling information between the two infrastructures. Signaling information contains data for connection control, signaling etc.
In ‘classical’—i.e. centralized communication devices switching is undertaken for example in accordance with the principle of a time division multiplex between communication terminals with fixed line assignments—the signaling protocol ‘DSS1’ is thus known. The DSS1 (‘Digital User System No. 1’) protocol is a European ISDN (‘Integrated Services Digital Network’) protocol based on ITU-T 1.411 (‘International Telecommunication Union’) for the signaling channel of the European Euro-ISDN.
In modern communication systems signaling information additionally includes data to support expanded service features, e.g. to display a name and further Information of a calling or called user at a communication terminal.
With the use of expansion modules—also known in technical circles as ‘gateways’—for ‘classical’ communication devices to connect VoIP communication terminals, the entire range of service features is not currently available either at wired or at VoIP communication terminals.